Member Avatar for Harvey-St

Hi people,

Another question,

We have a new VoIP Phone SPA525G2 with Zen Internet with the public IP address (Static IP) with a new Draytek 2830n router. The issue is, there are no incoming calls but I can make outgoing calls to any number on the phone fine. When you ring the number from your own mobile phone to the IP phone, you can hear it dialling on your mobile but no incoming call or activity on the VoIP Phone.

When I logged into the phone via the interface from the IP address under SIP it says it’s registered successfully so I am entering the SIP username and password from the broadband provider correctly including the proxy address and port numbers are 5060.

What configuration am I missing? I have opened ports 5060 on the Draytek and port forwarded them also to the phone private IP address. We only have 1 IP phone.

The Draytek router 2830n does not have VoIP features but someone said you don’t have to have a router that has VoIP. Would this be the issue.

May be JorgeM know of this ??

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Useful follow-ups to and — when a handset can place calls but never rings for incoming, the problem is almost always that the provider's INVITE never reaches the phone or the router/firewall mangles the SIP signalling. The quickest way to prove which side is at fault is a simple, reproducible trace: either capture the SIP packets on your LAN while someone dials in, or ask the carrier for a SIP trace showing the INVITE and the Contact/To headers they used.

Troubleshooting checklist (run these in order):

  1. Ask the provider for a SIP trace for an attempted inbound call. The trace will show whether they sent an INVITE and to which IP:port — that tells you immediately whether the call left their network correctly.
  2. While someone calls, run a packet capture on the LAN (Wireshark/tcpdump) or use the phone’s SIP/log page to see if an INVITE arrives. If the INVITE reaches your router but not the phone, the router is the blocker. If it never arrives at your router, the carrier must fix their routing.
  3. Inspect the phone’s registration/contact information (SIP status). If the registrar stored an unreachable contact (private RFC1918 address or wrong port), incoming INVITEs will fail even though outbound calls and registrations appear OK. If that’s the case, enable NAT-traversal measures (STUN or outbound-proxy/keepalives) on the phone or ask the carrier for guidance.
  4. Disable any SIP ALG / SIP helpers on every router in the path (consumer routers commonly rewrite SIP and break incoming calls). If you can, test by placing the phone in a DMZ or using a directly-attached device to eliminate router/NAT as the variable.

Additional notes: forwarding only UDP 5060 is often insufficient — RTP uses a separate dynamic UDP range, so if you choose port-forwarding instead of registration-based routing, forward the phone’s RTP range too. If these steps don’t isolate it, escalate to the carrier with your packet capture and the phone’s SIP logs so they can match their call attempt to what they sent.

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Member Avatar for CimmerianX

SIP does not like NAT. That's a fact. To get around this problem several solutions are available depending on what your devices offer. i.e. STUN, ICE, TURN, SIP Fixup are all available. You may need to enable one or more of these features on your phone or router for this to work.

Member Avatar for rubberman

Alternatively, you may need to reconfigure your router/firewall to allow incoming connections via ports 5060 and 5061. There may be other ports that you need to open as well, but I'm not sure if you need them, for example SIP directory services (port 5059).

Edited by rubberman
Member Avatar for Harvey-St

Thank you Cimmerian and Rubberman.

The VoIP provider which are also the broadband provider Zen Internet, said they don't use STUN etc etc, just the proxy address and register address which is voip.zen.co.uk and they said only port 5060 is used. They said no other settings used.

I have also tried Zen's own router (Thompson Techicolour) and also configured it on DMZ with their static public IP address direct without going through NAT or Firewall or Internal network correctly according to Zen. I did recall I tried all ports and opened all port then pointed it to the IP phone private IP address.

The provider must of not configured properly or forgotten to set or the Cisco phone not compatible.

Thanks

Member Avatar for simonrobert

For configuring any desktop VoIP phone, it is required to provide the following information:

Account Active: select 'Yes' (the existence of this setting depends on the type of the VoIP phone)
Account Name: e.g. your company
SIP Server: the IP address of your PBX
SIP User ID: the user part of an SIP address
Authenticate ID: **can be same or different from SIP User ID
**Authenticate Password: it is usually optional
**Name: ** it is usually optional, e.g. John Sample

(Source of information)

Edited by simonrobert
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